session-android/jni/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

55 lines
1.3 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_
#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
// forward declaration
struct GSMFR_encinst_t_;
struct GSMFR_decinst_t_;
namespace webrtc {
namespace acm2 {
class ACMGSMFR : public ACMGenericCodec {
public:
explicit ACMGSMFR(int16_t codec_id);
~ACMGSMFR();
// for FEC
ACMGenericCodec* CreateInstance(void);
int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
protected:
void DestructEncoderSafe();
int16_t InternalCreateEncoder();
void InternalDestructEncoderInst(void* ptr_inst);
int16_t EnableDTX();
int16_t DisableDTX();
GSMFR_encinst_t_* encoder_inst_ptr_;
};
} // namespace acm2
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_